Dialogic D/41ESC

Global SCSA 4-Port Voice Processing Board

Features and Benefits

• Four independent voice processing ports in a single PC ISA slot supports low- to medium-density voice systems
• Approved for use in numerous countries throughout North and South America, Europe, and Asia/Pacific
• SCbus connectivity enables applications requiring switching and allows access to additional resources such as fax, text-to-speech, and automatic speech recognition
• Supports Windows NT* and Windows 95*, including TAPI/WAVE*
• A-law or µ-law voice coding at dynamically selectable data rates, 24 Kb/s to 64 Kb/s, selectable on a channel-by-channel basis for optimal tradeoff between disk storage and voice quality
• Spring Ware downloadable signal and call processing firmware, provides easy feature enhancement and field-proven performance based on over 4 million installed ports
• International Caller ID capability via on-hook audio path. Supports Bellcore CLASS*, UK CLI, and other international protocols.
• DTMF (touch tone) detection provides reliable detection during voice playback-allows callers to "type-ahead" through menus
• Patented outbound call progress analysis monitors outgoing call status quickly and accurately
• Configure multiple boards in a single PC for easy and cost-effective system expansion and to build scalable systems from four to 64 ports
• C language application program interfaces (APIs) for MS-DOS*, UNIX*, OS/2*, Windows NT, and Windows 95 shorten your development cycle so you can get your applications to market faster
• Support for Global Dial Pulse Detection (DPD) pulse-to-tone conversion software
• Supports software-based speech technologies
• Supports PBX Expert, a utility that simplifies switch integration

Applications

• Voice messaging/auto attendant
• Interactive voice response
• Audiotex
• Inbound and outbound telemarketing
• Operator services
• Dictation
• Auto dialers
• Telecomputing servers
• Notification systems
• Online data entry/query

The D/41ESC voice processing board brings DSP-based call technology to the global marketplace. The interface circuitry of the D/41ESC is approvable for connection to analog networks in over 30 countries. (See your sales engineer for a list of the latest approvals.)

The D/41ESC provides four telephone line interface circuits for direct connection to analog loop start lines. A dual-processor architecture, comprising a DSP (digital signal processor) and a general purpose microprocessor, handles all telephony signaling and performs DTMF (touch tone) and audio/voice signal processing tasks. This architecture allows the board to run Spring Ware call processing firmware features, including selectable rate, high-quality voice coding with speed control, outstanding DTMF detection with cut-through, and advanced outbound call progress analysis.

Multiple D/41ESCs can be installed in a single PC chassis enabling system expansion up to 64 ports. For adding resources such as facsimile, speech recognition and text-to-speech, the D/41ESC provides an SCbus option or PCM Expansion Bus (PEB) option as well as an Analog Expansion Bus (AEB). With the D/41ESC you can create applications that allow hands-free speed dialing from cellular car phones, hands-free voice mail, and automatic dialing of spoken numbers or names. Complicated numeric menu systems can be reduced to a small set of user-friendly spoken commands.

Downloaded Spring Ware firmware algorithms, executed by the onboard DSP, provide variable voice coding at 24 and 32 Kb/s ADPCM, and 48 and 64 Kb/s µ-law or A-law PCM. Sampling rates and coding methods are selectable on a channel-by-channel basis. Applications may dynamically switch sampling rate and coding method to optimize data storage or voice quality as the need arises. Spring Ware firmware also provides reliable DTMF detection, DTMF cut-through, and talk off/play off suppression over a wide variety of telephone line conditions.

The D/41ESC board:

• Connects directly to analog loop start telephone lines
• Offers application-controlled call answering
• Detects touch tones
• Plays voice messages to a caller
• Digitizes, compresses and records voice signals; places outbound calls and automatically monitors their progress
• All in real time on four independent channels.

Configurations

Use the D/41ESC board to build sophisticated computer telephony systems to which capabilities such as speech recognition, facsimile, and text-to-speech can be added. The D/41ESC shares a common hardware and firmware architecture with other Intel® Dialogic® SCbus, PEB, and AEB-based boards for maximum flexibility and scalability. Features can be added and systems can grow while protecting investment in hardware and application code. With only minimum modifications, applications can be easily ported to lower or higher-line-density platforms.

The D/41ESC installs in IBM PC AT (ISA bus) and compatible computers (80386, 80486, or Pentium® processor-based PC platforms). The D/41ESC provides for building integrated voice solutions scalable from four ports to 64 ports. The maximum number of lines that can be supported is dependent on the application, the amount of disk I/O required, and the host computer CPU and power supply.

Applications developed to run on the Proline/2V, DIALOG/4, D/41D, or D/41H family will run on a similar D/41ESC configuration. Developers can choose from a selection of products to build scalable, reliable, and economical computer telephony.

The D/41ESC is supported by System Software and Software Development Kits for many popular operating systems, including MS-DOS, OS/2, UNIX, Windows NT and Windows 95. These packages contain a set of tools for developing complex multichannel applications.

Functional Description

The D/41ESC uses a dual processor architecture that combines the signal processing capabilities of a DSP with the decision making and data movement functionality of a general purpose 80186 control microprocessor. This dual processor approach offloads many low level decision making tasks from the host computer and thus enables easier development of more powerful applications. This architecture handles real time events, manages data flow to the host PC for faster system response time, reduces host PC processing demands, processes DTMF and telephony signaling, and frees the DSP to perform signal processing on the incoming call.

Each of four analog loop start telephone line interfaces on the D/41ESC receives analog voice and telephony signaling information from the telephone network (see block diagram). Each telephone line interface uses reliable, solid state hook switches (no mechanical contacts), and FCC-part 68 class B ring detection circuitry. This FCC-approved ring detector is less susceptible to spurious rings created by random voltage fluctuations on the network. Each interface also incorporates circuitry that protects against high voltage spikes and adverse network conditions and allows applications to go off hook any time during ring cadence without damaging the board.

Inbound telephony signaling (ring detection, loop current detection, and Caller ID information) is conditioned by the line interface and routed via a control bus to the control processor. The control processor responds to these signals, informs the application of telephony signaling status, and instructs the line interface to transmit outbound signaling (on-hook/off-hook) to the telephone network.

The audio voice signal from the network is bandpass filtered and conditioned by the line interface and then applied to a CODEC (COder/DECoder) circuit. The CODEC filters, samples, and digitizes the inbound analog audio signal and passes this digitized audio signal to a Motorola* DSP.

Based on Spring Ware firmware loaded in DSP SRAM, the DSP performs the following signal analysis and operations on this incoming data:

• Applies automatic gain control to compensate for variations in the level of the incoming audio signal
• Applies an Adaptive Differential Pulse Code Modulation (ADPCM) or Pulse Code Modulation (PCM) algorithm to compress the digitized voice and save disk storage space
• Detects the presence of tones - DTMF, MF, or an application-defined single or dual tone
• Detects silence to determine whether the line is quiet and the caller is not responding

• Expands stored, compressed audio data for playback
• Adjusts the volume and rate of speed of playback upon application or user request
• Generates tones - DTMF, MF, or any application-defined general purpose tone

• Transmits an off-hook signal to the telephone network
• Dials out (makes an outbound call)
• Monitors and reports results: - line busy or congested
  - operator intercept
  - ring
  - no answer
  - if answered, whether answered by a person, an answering machine, a fax machine, or a modem
  

The D/41ESC also supports Global Dial Pulse Detection (DPD) Software that recognizes dial pulse digits even in the most difficult telephony environments.

When recording speech, the DSP can use different digitizing rates from 24 to 64 Kb/s as selected by the application for the best speech quality and most efficient storage. The digitizing rate is selected on a channel-by-channel basis and can be changed each time a record or play function is initiated. The DSP-processed speech is transmitted by the control processor to the host PC for disk storage. When replaying a stored file, the processor retrieves the voice information from the host PC and passes it to the DSP, which converts the file into digitized voice. The DSP sends digitized voice and appropriate signaling responses to the CODEC to be converted into analog format for transmission to the telephone network.

Signaling data (on-/off-hook, ringing, Caller ID, etc.) is passed to the onboard control processor and transmitted to the application via a dual-port shared RAM and the host PC ISA bus.

When using the D/41ESC board with SCbus or PEB, digital voice and signaling information from a network board or other resource enter the board via the SCbus interface. These signals are managed by a SC2000 chip that acts as the traffic coordinator and matrix switch to buffer the high-speed digital data from the bus until the data for each channel can be transmitted to the DSP.

The SC2000 chip transmits several lower speed data streams over a single high-speed channel, either the SCbus or the PEB. The bus configuration is set when the firmware is downloaded at system initialization. This chip incorporates matrix-switching capabilities. Under control of the onboard control processor, the SC2000 chip can connect any call being processed to any of the four analog lines or to any SCbus or PEB time slot (1024 for the SCbus; 24 for the PEB in T-1 mode, or 32 in E-1 mode). This enables the application to switch calls to or from other resources, such as facsimile or speech recognition, as they are needed, or to reroute calls.

The SC2000 chip can bundle time slots to carry high bandwidth data and can broadcast to multiple resources over the SCbus.

The onboard microprocessor controls all operations of the D/41ESC via a local bus and interprets and executes commands from the host PC. This microprocessor handles real-time events, manages data flow to the host PC to provide faster system response time, reduces PC host processing demands, processes DTMF and telephony signals before passing them to the application, and frees the DSP to perform signal processing. Communications between this microprocessor and the host PC is via the dual port shared RAM that acts as an input/output buffer and thus increases the efficiency of disk file transfers. This RAM interfaces to the host PC via the AT® (ISA) bus. All operations are interrupt driven to meet the demands of real-time systems. When the system is initialized, Spring Ware firmware to control all board operations is downloaded from the host PC to the onboard code/data RAM and DSP SRAM. This downloadable firmware gives the board all of its intelligence and enables easy feature enhancement and upgrades.

The Board Locator Technology circuit operates in conjunction with a rotary switch to determine and set nonconflicting PC memory and IRQ interrupt level parameters. This feature eliminates the need to set confusing jumpers or DIP switches.

D/41ESC Technical Specifications**

D/41ESC Spring Ware Firmware Technical Specifications**

*All company names, products, and services mentioned are the trademarks or registered trademarks of their respective owners.

** All specifications are subject to change without notice.

† Analog levels: 0 dBm0 corresponds to a level of +3 dBm at tip-ring analog point. Values vary depending on country requirements; contact your Intel Sales Engineer.

‡ Average speech mandates +16 dB peaks above average and preserves -13 dB valleys below average.

Hardware System Requirements

• Intel386™, Intel486™, or Pentium® processor-based, IBM PC AT (ISA) bus or compatible computer. Operating system hardware requirements vary according to the number of channels being used.